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Showing posts with the label SIP

In and out of WEBRTC - All in one blog

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                                            WEBRTC WebRTC is a great technology, but it may not fit everywhere and you should understand what you are doing if you decided to use it. WebRTC is very appealing as a vehicle to extend the reach of IMS services. WebRTC is available in five popular browsers in production state: Google Chrome; Opera; Mozilla Firefox; Chrome Android; Firefox Android.  WORKING WITH WEBRTC :-                                                       1. GetuserMedia  API It access the camera and microphone of devices -------------------------------------------- 2. Peer connection API It exchanges various protocols for transmitting data/video from one peer to another peer...

Set up your VOIP based SIP soft phone and know more about VOIP

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VOIP (Voice over internet protocol)        The  trending technology of present and near future. I have spent months of efforts on voip , its a huge topic so to minimize your time and also to refresh my voip skills , I have decided to write this blog. Main Topics covered :- SIP (session initiation protocol) ASTERISK (open source voip server platform)     SIP is a part of Application layer (fig1) in TCP/IP , its major function is to initiate/terminate a sip session and transfer it to TCP/UDP for transmission. In simple terms it holds the FROM and TO addresses in a post , the rest is carried using TCP/IP protocol.  fig1 : SIP in TCP/IP The basic operation of SIP is an INVITE and ACK via proxy, once the opposite Sip phone B accepts the call, a media session is established. Its simple as shown in fig 2. The proxy might be a VOIP services provided by certain websites such as   www.linphone.org ,  SIP2SIP.info  etc or it...